U.S. patent application number 10/125663 was filed with the patent office on 2002-12-19 for in-situ transducer modeling in a digital hearing instrument.
Invention is credited to Armstrong, Stephen W..
Application Number | 20020191800 10/125663 |
Document ID | / |
Family ID | 23092269 |
Filed Date | 2002-12-19 |
United States Patent
Application |
20020191800 |
Kind Code |
A1 |
Armstrong, Stephen W. |
December 19, 2002 |
In-situ transducer modeling in a digital hearing instrument
Abstract
A method for in-situ transducer modeling in a digital hearing
instrument is provided. In one embodiment, a personal computer is
coupled to a processing device in the digital hearing instrument
and configures the processing device to operate as a level detector
and a tone generator. An audio signal generated by the personal
computer is received by a microphone-under-test (MUT) in the
digital hearing instrument and the energy level of the received
audio signal is determined by the level detector. In addition, an
audio output signal generated by the tone generator and a
speaker-under-test (SUT) in the digital hearing instrument is
received by a microphone, and the energy level of the audio output
signal is determined by a level meter. The energy levels of the
received audio signal and the audio output signal are used by the
personal computer to generate an electro-acoustic model of the
digital hearing instrument.
Inventors: |
Armstrong, Stephen W.;
(Burlington, CA) |
Correspondence
Address: |
Joseph M. Sauer, Esq.
Jones, Day, Reavis & Pogue
North Point
901 Lakeside Avenue
Cleveland
OH
44114
US
|
Family ID: |
23092269 |
Appl. No.: |
10/125663 |
Filed: |
April 18, 2002 |
Related U.S. Patent Documents
|
|
|
|
|
|
Application
Number |
Filing Date |
Patent Number |
|
|
60284984 |
Apr 19, 2001 |
|
|
|
Current U.S.
Class: |
381/60 |
Current CPC
Class: |
H04R 25/30 20130101;
H04R 29/001 20130101; H04R 25/70 20130101; H04R 29/004 20130101;
H04R 25/505 20130101 |
Class at
Publication: |
381/60 |
International
Class: |
H04R 029/00 |
Claims
I claim:
1. A method of in-situ transducer modeling in a digital hearing
instrument, comprising the steps of: providing a
microphone-under-test (MUT) coupled to a level detector in the
digital hearing instrument; generating an audio signal using a
personal computer coupled to a tone generator; receiving the audio
signal with the MUT in the digital hearing instrument; determining
the energy level of the received audio signal using the level
detector in the digital hearing instrument; coupling the personal
computer to the level detector through an external port connection
in the digital hearing instrument; recording the energy level of
the received audio signal with the personal computer; and
developing an electro-acoustic model of the digital hearing
instrument using the recorded energy level of the received audio
signal.
2. The method of claim 1, comprising the additional step of:
configuring a processing device in the digital hearing instrument
to operate as the level detector.
3. The method of claim 1, comprising the additional steps of:
providing a speaker-under-test (SUT) coupled to an internal tone
generator in the digital hearing instrument; generating an audio
output signal with the internal tone generator and SUT; receiving
the audio output signal with a microphone; determining the energy
level of the audio output signal with a level meter; recording the
energy level of the audio output signal with the personal computer;
and developing the electro-acoustic model of the digital hearing
instrument using the recorded energy level of the audio output
signal.
4. The method of claim 3, comprising the additional steps of:
coupling the personal computer to a processing device in the
digital hearing instrument; and configuring the processing device
in the digital hearing instrument to operate as the internal tone
generator.
5. A method of in-situ transducer modeling in a digital hearing
instrument, comprising the steps of: a microphone-under-test (MUT)
and a speaker-under-test (SUT) in the digital hearing instrument;
generating an audio signal using a personal computer coupled to a
tone generator; receiving the audio signal with the MUT; coupling
the personal computer to a processing device in the digital hearing
instrument; configuring the processing device to operate as a level
detector; determining the energy level of the received audio signal
using the level detector; applying a gain to the received audio
signal to generate an amplified audio signal; determining the
energy level of the amplified audio signal using the level
detector; using the personal computer to determine a difference
between the energy levels of the received and amplified audio
signals; determining if the difference between the energy levels of
the received and amplified audio signals meets a pre-determined
hearing aid characteristic; and if the difference between the
energy levels of the received and amplified audio signals does not
meet the predetermined hearing aid characteristic, then adjusting
the gain applied to the received audio signal.
Description
CROSS-REFERENCE TO RELATED APPLICATION
[0001] This application claims priority from and is related to the
following prior application: In-Situ Transducer Modeling In a
Digital Hearing Instrument, U.S. Provisional Application No.
60/284,984, filed Apr. 19, 2001. In addition, this application is
related to the following co-pending application which is owned by
the assignee of the present invention: Digital Hearing Aid System,
United States Patent Application [application number not yet
available], filed Apr. 12, 2001. These prior applications,
including the entire written descriptions and drawing figures, are
hereby incorporated into the present application by reference.
BACKGROUND
[0002] 1. Field of the Invention
[0003] This invention generally relates to digital hearing
instruments. More specifically, the invention provides a method in
a digital hearing instrument for in-situ modeling of the instrument
transducers (i.e., microphone(s) and speaker(s)) using the digital
hearing instrument as a signal processor.
[0004] 2. Description of the Related Art
[0005] Digital hearing instruments are known in this field. These
instruments typically include a plurality of transducers, including
at least one microphone and at least one speaker. Some instruments
include a plurality of microphones, such as a front microphone and
a rear microphone to provide directional hearing.
[0006] Hearing aid fitting software is often used during the
customization of such instruments in order to configure the
instrument settings for a particular user. This software typically
presents information regarding the instrument to the fitting
operator in the form of graphs displayed on a personal computer.
The graphs are intended to display the performance of the
instrument given the current settings of the device. In order to
display these performance graphs, the fitting software requires
mathematical models of the electrical transfer function of the
instrument in conjunction with electro-acoustical models of the
microphone and the speaker.
[0007] Traditionally, the electro-acoustical models of the
microphone and the speaker are derived independently from the
fitting process by skilled technicians. FIG. 2 is a block diagram
showing the traditional method of characterizing a microphone in a
digital hearing instrument. Here, the microphone-under-test (MUT)
is coupled to a meter 108 for measuring the voltage output from the
microphone. This measured voltage is applied to a custom test and
measurement system 104, which is also coupled to a tone generator
106 and an external speaker 110. Operationally, the test and
measurement system 104 controls the tone generator 106 and causes
it to sweep across a particular frequency range of interest, during
which time it takes measurement data from the meter 108. The test
an measurement system then derives an electro-acoustical model 112
of the MUT 102 using the data gathered from the meter 108.
[0008] FIG. 3 is a block diagram showing the traditional method of
characterizing a speaker in a digital hearing instrument. Here, the
speaker-under-test (SUT) is coupled to the tone generator 106. The
test and measurement system 104 causes the tone generator 106 to
drive the SUT with a known signal level while the acoustic sound
pressure developed from the SUT is quantified by a test microphone
102 and level meter 108. Using the data gathered from the level
meter 108, the test and measurement system 104 then derives the
electro-acoustical model for the SUT 110.
[0009] The problem with the foregoing traditional characterization
and modeling methods is that the specialized equipment required to
derive the models, i.e., the test and measurement system 104 and
other equipment, is very expensive, and also requires a skilled
technical operator.
BRIEF DESCRIPTION OF THE DRAWINGS
[0010] FIG. 1 is a block diagram of an exemplary digital hearing
instrument including a plurality of transducers;
[0011] FIG. 2 is a block diagram showing the traditional method of
characterizing a microphone in a digital hearing instrument;
[0012] FIG. 3 is a block diagram showing the traditional method of
characterizing a speaker in a digital hearing instrument;
[0013] FIG. 4 is a block diagram showing a method of in-situ
transducer modeling according to the present invention; and
[0014] FIG. 5 is a block diagram showing another method of in-situ
transducer modeling according to the present invention.
SUMMARY
[0015] A method for in-situ transducer modeling in a digital
hearing instrument is provided. In one embodiment, a personal
computer is coupled to a processing device in the digital hearing
instrument and configures the processing device to operate as a
level detector and an internal tone generator. An audio signal
generated by the personal computer is received by a
microphone-under-test (MUT) in the digital hearing instrument and
the energy level of the received audio signal is determined by the
level detector. In addition, an audio output signal generated by
the tone generator and a speaker-under-test (SUT) in the digital
hearing instrument is received by a microphone, and the energy
level of the audio output signal is determined by a level meter.
The energy levels of the received audio signal and the audio output
signal are used by the personal computer to generate an
electro-acoustic model of the digital hearing instrument.
[0016] In another embodiment, the personal computer configures the
processing device in the digital hearing instrument to operate as a
level detector. An audio signal generated by the personal computer
is received by a MUT in the digital hearing instrument, and the
energy level of the received audio signal is determined by the
level detector. A gain is then applied to the received audio
signal, and the energy level of the amplified audio signal is
determined by the level detector. The personal computer compares
the energy levels of the received and amplified audio signals and
adjusts the gain such that the digital hearing instrument meets
predetermined hearing aid characteristics.
DETAILED DESCRIPTION OF THE DRAWINGS
[0017] Turning now to the drawing figures, FIG. 1 is a block
diagram of an exemplary digital hearing aid system 12. The digital
hearing aid system 12 includes several external components 14, 16,
18, 20, 22, 24, 26, 28, and, preferably, a single integrated
circuit (IC) 12A. The external components include a pair of
microphones 24, 26, a tele-coil 28, a volume control potentiometer
24, a memory-select toggle switch 16, battery terminals 18, 22, and
a speaker 20.
[0018] Sound is received by the pair of microphones 24, 26, and
converted into electrical signals that are coupled to the FMIC 12C
and RMIC 12D inputs to the IC 12A. FMIC refers to "front
microphone," and RMIC refers to "rear microphone." The microphones
24, 26 are biased between a regulated voltage output from the RREG
and FREG pins 12B, and the ground nodes FGND 12F and RGND 12G. The
regulated voltage output on FREG and RREG is generated internally
to the IC 12A by regulator 30.
[0019] The tele-coil 28 is a device used in a hearing aid that
magnetically couples to a telephone handset and produces an input
current that is proportional to the telephone signal. This input
current from the tele-coil 28 is coupled into the rear microphone
A/D converter 32B on the IC 12A when the switch 76 is connected to
the "T" input pin 12E, indicating that the user of the hearing aid
is talking on a telephone. The tele-coil 28 is used to prevent
acoustic feedback into the system when talking on the
telephone.
[0020] The volume control potentiometer 14 is coupled to the volume
control input 12N of the IC. This variable resistor is used to set
the volume sensitivity of the digital hearing aid.
[0021] The memory-select toggle switch 16 is coupled between the
positive voltage supply VB 18 and the memory-select input pin 12L.
This switch 16 is used to toggle the digital hearing aid system 12
between a series of setup configurations. For example, the device
may have been previously programmed for a variety of environmental
settings, such as quiet listening, listening to music, a noisy
setting, etc. For each of these settings, the system parameters of
the IC 12A may have been optimally configured for the particular
user. By repeatedly pressing the toggle switch 16, the user may
then toggle through the various configurations stored in the
read-only memory 44 of the IC 12A.
[0022] The battery terminals 12K, 12H of the IC 12A are preferably
coupled to a single 1.3 volt zinc-air battery. This battery
provides the primary power source for the digital hearing aid
system.
[0023] The last external component is the speaker 20. This element
is coupled to the differential outputs at pins 12J, 121 of the IC
12A, and converts the processed digital input signals from the two
microphones 24, 26 into an audible signal for the user of the
digital hearing aid system 12.
[0024] There are many circuit blocks within the IC 12A. Primary
sound processing within the system is carried out by a sound
processor 38 and a directional processor and headroom expander 50.
A pair of A/D converters 32A, 32B are coupled between the front and
rear microphones 24, 26, and the directional processor and headroom
expander 50, and convert the analog input signals into the digital
domain for digital processing. A single D/A converter 48 converts
the processed digital signals back into the analog domain for
output by the speaker 20. Other system elements include a regulator
30, a volume control A/D 40, an interface/system controller 42, an
EEPROM memory 44, a power-on reset circuit 46, a oscillator/system
clock 36, a summer 71, and an interpolator and peak clipping
circuit 70.
[0025] The sound processor 38 preferably includes a pre-filter 52,
a wide-band twin detector 54, a band-split filter 56, a plurality
of narrow-band channel processing and twin detectors 58A-58D, a
summation block 60, a post filter 62, a notch filter 64, a volume
control circuit 66, an automatic gain control output circuit 68, a
squelch circuit 72, and a tone generator 74.
[0026] Operationally, the digital hearing aid system 12 processes
digital sound as follows. Analog audio signals picked up by the
front and rear microphones 24, 26 are coupled to the front and rear
A/D converters 32A, 32B, which are preferably Sigma-Delta
modulators followed by decimation filters that convert the analog
audio inputs from the two microphones into equivalent digital audio
signals. Note that when a user of the digital hearing aid system is
talking on the telephone, the rear A/D converter 32B is coupled to
the tele-coil input "T" 12E via switch 76. Both the front and rear
A/D converters 32A, 32B are clocked with the output clock signal
from the oscillator/system clock 36 (discussed in more detail
below). This same output clock signal is also coupled to the sound
processor 38 and the D/A converter 48.
[0027] The front and rear digital sound signals from the two A/D
converters 32A, 32B are coupled to the directional processor and
headroom expander 50. The rear A/D converter 32B is coupled to the
processor 50 through switch 75. In a first position, the switch 75
couples the digital output of the rear A/D converter 32 B to the
processor 50, and in a second position, the switch 75 couples the
digital output of the rear A/D converter 32B to summation block 71
for the purpose of compensating for occlusion.
[0028] Occlusion is the amplification of the users own voice within
the ear canal. The rear microphone can be moved inside the ear
canal to receive this unwanted signal created by the occlusion
effect. The occlusion effect is usually reduced by putting a
mechanical vent in the hearing aid. This vent, however, can cause
an oscillation problem as the speaker signal feeds back to the
microphone(s) through the vent aperture. Another problem associated
with traditional venting is a reduced low frequency response
(leading to reduced sound quality). Yet another limitation occurs
when the direct coupling of ambient sounds results in poor
directional performance, particularly in the low frequencies. The
system shown in FIG. 1 solves these problems by canceling the
unwanted signal received by the rear microphone 26 by feeding back
the rear signal from the A/D converter 32B to summation circuit 71.
The summation circuit 71 then subtracts the unwanted signal from
the processed composite signal to thereby compensate for the
occlusion effect.
[0029] The directional processor and headroom expander 50 includes
a combination of filtering and delay elements that, when applied to
the two digital input signals, form a single,
directionally-sensitive response. This directionally-sensitive
response is generated such that the gain of the directional
processor 50 will be a maximum value for sounds coming from the
front microphone 24 and will be a minimum value for sounds coming
from the rear microphone 26.
[0030] The headroom expander portion of the processor 50
significantly extends the dynamic range of the A/D conversion,
which is very important for high fidelity audio signal processing.
It does this by dynamically adjusting the operating points of the
A/D converters 32A/32B. The headroom expander 50 adjusts the gain
before and after the A/D conversion so that the total gain remains
unchanged, but the intrinsic dynamic range of the A/D converter
block 32A/32B is optimized to the level of the signal being
processed.
[0031] The output from the directional processor and headroom
expander 50 is coupled to the pre-filter 52 in the sound processor
38, which is a general-purpose filter for pre-conditioning the
sound signal prior to any further signal processing steps. This
"pre-conditioning" can take many forms, and, in combination with
corresponding "post-conditioning" in the post filter 62, can be
used to generate special effects that may be suited to only a
particular class of users. For example, the pre-filter 52 could be
configured to mimic the transfer function of the user's middle ear,
effectively putting the sound signal into the "cochlear domain."
Signal processing algorithms to correct a hearing impairment based
on, for example, inner hair cell loss and outer hair cell loss,
could be applied by the sound processor 38. Subsequently, the
post-filter 62 could be configured with the inverse response of the
pre-filter 52 in order to convert the sound signal back into the
"acoustic domain" from the "cochlear domain." Of course, other
pre-conditioning/post-conditionin- g configurations and
corresponding signal processing algorithms could be utilized.
[0032] The pre-conditioned digital sound signal is then coupled to
the band-split filter 56, which preferably includes a bank of
filters with variable comer frequencies and pass-band gains. These
filters are used to split the single input signal into four
distinct frequency bands. The four output signals from the
band-split filter 56 are preferably in-phase so that when they are
summed together in summation block 60, after channel processing,
nulls or peaks in the composite signal (from the summation block)
are minimized.
[0033] Channel processing of the four distinct frequency bands from
the band-split filter 56 is accomplished by a plurality of channel
processing/twin detector blocks 58A-58D. Although four blocks are
shown in FIG. 1, it should be clear that more than four (or less
than four) frequency bands could be generated in the band-split
filter 56, and thus more or less than four channel processing/twin
detector blocks 58 may be utilized with the system.
[0034] Each of the channel processing/twin detectors 58A-58D
provide an automatic gain control ("AGC") function that provides
compression and gain on the particular frequency band (channel)
being processed. Compression of the channel signals permits quieter
sounds to be amplified at a higher gain than louder sounds, for
which the gain is compressed. In this manner, the user of the
system can hear the full range of sounds since the circuits 58A-58D
compress the full range of normal hearing into the reduced dynamic
range of the individual user as a function of the individual user's
hearing loss within the particular frequency band of the
channel.
[0035] The channel processing blocks 58A-58D can be configured to
employ a twin detector average detection scheme while compressing
the input signals. This twin detection scheme includes both slow
and fast attack/release tracking modules that allow for fast
response to transients (in the fast tracking module), while
preventing annoying pumping of the input signal (in the slow
tracking module) that only a fast time constant would produce. The
outputs of the fast and slow tracking modules are compared, and the
compression parameters are then adjusted accordingly. The
compression ratio, channel gain, lower and upper thresholds (return
to linear point), and the fast and slow time constants (of the fast
and slow tracking modules) can be independently programmed and
saved in memory 44 for each of the plurality of channel processing
blocks 58A-58D.
[0036] FIG. 1 also shows a communication bus 59, which may include
one or more connections for coupling the plurality of channel
processing blocks 58A-58D. This inter-channel communication bus 59
can be used to communicate information between the plurality of
channel processing blocks 58A-58D such that each channel (frequency
band) can take into account the "energy" level (or some other
measure) from the other channel processing blocks. Preferably, each
channel processing block 58A-58D would take into account the
"energy" level from the higher frequency channels. In addition, the
"energy" level from the wide-band detector 54 may be used by each
of the relatively narrow-band channel processing blocks 58A-58D
when processing their individual input signals.
[0037] After channel processing is complete, the four channel
signals are summed by summation bock 60 to form a composite signal.
This composite signal is then coupled to the post-filter 62, which
may apply a post-processing filter function as discussed above.
Following post-processing, the composite signal is then applied to
a notch-filter 64, that attenuates a narrow band of frequencies
that is adjustable in the frequency range where hearing aids tend
to oscillate. This notch filter 64 is used to reduce feedback and
prevent unwanted "whistling" of the device. Preferably, the notch
filter 64 may include a dynamic transfer function that changes the
depth of the notch based upon the magnitude of the input
signal.
[0038] Following the notch filter 64, the composite signal is
coupled to a volume control circuit 66. The volume control circuit
66 receives a digital value from the volume control A/D 40, which
indicates the desired volume level set by the user via
potentiometer 14, and uses this stored digital value to set the
gain of an included amplifier circuit.
[0039] From the volume control circuit, the composite signal is
coupled to the AGC-output block 68. The AGC-output circuit 68 is a
high compression ratio, low distortion limiter that is used to
prevent pathological signals from causing large scale distorted
output signals from the speaker 20 that could be painful and
annoying to the user of the device. The composite signal is coupled
from the AGC-output circuit 68 to a squelch circuit 72, that
performs an expansion on low-level signals below an adjustable
threshold. The squelch circuit 72 uses an output signal from the
wide-band detector 54 for this purpose. The expansion of the
low-level signals attenuates noise from the microphones and other
circuits when the input S/N ratio is small, thus producing a lower
noise signal during quiet situations. Also shown coupled to the
squelch circuit 72 is a tone generator block 74, which is included
for calibration and testing of the system.
[0040] The output of the squelch circuit 72 is coupled to one input
of summation block 71. The other input to the summation bock 71 is
from the output of the rear A/D converter 32B, when the switch 75
is in the second position. These two signals are summed in
summation block 71, and passed along to the interpolator and peak
clipping circuit 70. This circuit 70 also operates on pathological
signals, but it operates almost instantaneously to large peak
signals and is high distortion limiting. The interpolator shifts
the signal up in frequency as part of the D/A process and then the
signal is clipped so that the distortion products do not alias back
into the baseband frequency range.
[0041] The output of the interpolator and peak clipping circuit 70
is coupled from the sound processor 38 to the D/A H-Bridge 48. This
circuit 48 converts the digital representation of the input sound
signals to a pulse density modulated representation with
complimentary outputs. These outputs are coupled off-chip through
outputs 12J, 12I to the speaker 20, which low-pass filters the
outputs and produces an acoustic analog of the output signals. The
D/A H-Bridge 48 includes an interpolator, a digital Delta-Sigma
modulator, and an H-Bridge output stage. The D/A H-Bridge 48 is
also coupled to and receives the clock signal from the
oscillator/system clock 36 (described below).
[0042] The interface/system controller 42 is coupled between a
serial data interface pin 12M on the IC 12, and the sound processor
38. This interface is used to communicate with an external
controller for the purpose of setting the parameters of the system.
These parameters can be stored on-chip in the EEPROM 44. If a
"black-out" or "brown-out" condition occurs, then the power-on
reset circuit 46 can be used to signal the interface/system
controller 42 to configure the system into a known state. Such a
condition can occur, for example, if the battery fails.
[0043] FIG. 4 is a block diagram showing a method of in-situ
transducer modeling according to one embodiment of the present
invention. Here, instead of the specialized test and measurement
system 104 used in the traditional characterization and modeling
methods, a personal computer 128 is substituted. The personal
computer 128 is coupled to a tone generator 106 and a level meter
108. The personal computer 128 is also coupled to the digital
hearing instrument 12 via an external port connection 130, such as
a serial port.
[0044] Within the digital hearing instrument is the
microphone-under-test (MUT) 102 and the speaker-under-test (SUT)
120. Also included in the digital hearing instrument is a
processing device, such as a programmable digital signal processor
(DSP) 122. This processing device 122 may be similar to sound
processor 38 shown in FIG. 1.
[0045] Software operating on the personal computer 128 configures
the DSP 122 to operate as a level detector (LD) 124 for incoming
MUT 102 signals, and as an internal tone generator (TG) 126 for the
SUT 120. This software then performs the required frequency sweep
measurements using the external speaker 110 and the MUT/LD
combination 102/124 within the digital hearing instrument 12. The
software also performs the frequency sweep of the TG/SUT
combination 126/120 and measures with the external microphone 122
and level meter 108. By configuring the DSP 122 in this manner, the
personal computer can replace the more complicated test and
measurement system 104 shown in FIGS. 2 and 3, and enables a
non-skilled operator to generate the electro-acoustic models 112 of
the digital hearing instrument 12.
[0046] FIG. 5 is a block diagram showing another method of in-situ
transducer modeling according to the present invention. In this
method, the processing device 122 does not include a tone generator
(TG) 126. Instead, the TG 126 function is achieved by using the
external speaker 110 transduced by the MUT 102, and by adjusting
the gain of the circuit so that the signal level presented to the
SUT 120, and measured by an additional level detector 124, meets
the pre-determined hearing instrument characteristics. Again, the
software operating at the personal computer 128 performs the
desired frequency sweep with the additional step of adjusting the
gain at each frequency step.
[0047] This written description uses examples to disclose the
invention, including the best mode, and also to enable any person
skilled in the art to make and use the invention. The patentable
scope of the invention is defined by the claims, and may include
other examples that occur to those skilled in the art.
* * * * *